The invention relates to telecommunications systems using packet transmission and radiocommunication transfer. More particularly, it is related to a method for allocating packets coming from a packet transmission within a limited timeslot in a radiocommunication system.
In transmission systems that aim at offering interactive services, it is crucial to define a service quality required for the communication. Since these services are linked with the time response of all the systems, the time delays have to be optimized to ensure this service quality. It is known that there is no perception of time delay when system response times are below a limit defined by the type of service required. For example, phone communications have a limit of about 400 ms whereas medical remote systems have a limit of 5 ms. The time delay due to the system itself is then crucial.
In telecommunication and radiocommunication systems, the Bit Error Rate (BER) measuring the quality of the transmission, can be improved by using an error correcting code. In a packet (or cell) transmission, this correcting code can be used in two ways: an individual packet coding or a group coding with packet interleaving. The interleaving method allows reducing the Signal to Noise Ratio (SNR) threshold necessary for reaching the requested BER. Meanwhile, this method has to wait for the whole interleaved packet group in order to decode it.
In U.S. Pat. No. 5,231,633, a queueing and dequeueing mechanism for use in an integrated fast packet network, wherein fast packets from differing traffic types are multiplexed with one another through use of a weighted round-robin bandwidth allocation mechanism. Fast packets within a particular traffic type are selected for transmission through use of a head of line priority service (514), a packet discard mechanism, or both. The weighted round-robin bandwidth allocation mechanism functions, in part, based upon a credit counter for each queue group that represents a particular traffic type.
In U.S. Pat. No. 5,905,730, a packet scheduler is disclosed which provides a high degree of fairness in scheduling packets associated with different sessions. The scheduler also minimizes packet delay for packet transmission from a plurality of sessions which may have different requirements and may operate at different transfer rates. When a packet is received by the scheduler, the packet is assigned its own packet virtual start time based on whether the session has any pending packets and the values of the virtual finish time of the previous packet in the session and the packets arrival time. The scheduler then determines a virtual finish time of the packet by determining the transfer time required for the packet based upon its length and rate and by adding the transfer time to the packet virtual start time of the packet. The packet with the smallest virtual finish time is then scheduled for transfer. By selecting packets for transmission in the above described manner, the available bandwidth may be shared in pro-rata proportion to the guaranteed session rate, thereby providing a scheduler with a high degree of fairness while also minimizing the amount of time a packet waits in the scheduler before being served.
In U.S. Pat. No. 5,917,822, a method in accordance with the invention allocates bandwidth, fairly and dynamically, in a shared-media packet switched network to accommodate both elastic and inelastic applications. The method, executed by or in a head-end controller, allocates bandwidth transmission slots, converting requests for bandwidth into virtual scheduling times for granting access to the shared media. The method can use a weighted fair queuing algorithm or a virtual clock algorithm to generate a sequence of upstream slot/transmission assignment grants. The method supports multiple quality of service (QoS) classes via mechanisms which give highest priority to the service class with the most stringent QoS requirements.
These systems allow faster packet transmission, with a quality of service, but do not take into consideration the terminal characteristics. They are not fit to a telecommunication system that has a limited power ability, which is one of the problems solved by the invention.
In a Code Division Multiple Access (CDMA), Time Division Multiple Access (TDMA) or frequency division Multiple Access (FDMA) system using packet transmission, e.g. Asynchronous Transfer Mode (ATM), the communication with a terminal can be sporadic. The power necessary to enable the transmission to the terminal is adjusted following the propagation conditions. These accesses lead to a statistical multiplexing in passband and in power that is controlled by the filling algorithm. The interleaving coding may cause a rise of the time transfer of packets in a given terminal, which is not acceptable if a good service quality is required.
Resource allocation may be difficult with interleaving coding for the following reasons:                Service quality is required, especially for packet transfer;        Power is identical for the packets belonging to the same group;        The system should optimize its power and band consumption and avoid complementary or stuffing packets;        
In order to solve these problems, it is possible to use packet interleaving in a mono- or multi-terminal, a priori or a posteriori packet choice, fixed or variable position in the interleaved packets frame or appropriate carrier filling algorithm.
The first way to define the classes is based on the a priori knowledge of the emission power for each terminal, independently of the allocation process. At the end of the allocation, packets are interleaved in clusters with the same length as the interleaving.
The second way builds the classes after the allocation process, which correspond to an optimal a posteriori class definition. The loss due to power classes is minimized. An additional loss may also occur when the number of allocated packets is not a multiple of the interleaving length.
It is known that a mono-terminal requires a sufficient number of transmitted packets for an efficient coding in a limited time. The two methods can be combined, depending on the number of packets to be coded. The default method is the interleaving, except when the number of packets is too low. In that case another coding method is used. This solution is not really efficient and needs two decoders in the terminal.